diff --git a/tools/audio/av.py b/tools/audio/av.py
index 2c14baf9e..333b423d6 100644
--- a/tools/audio/av.py
+++ b/tools/audio/av.py
@@ -42,14 +42,16 @@ def wav2(i: BytesIO, o: BufferedWriter, format: str):
 
 def load_audio(
         file: Union[str, BytesIO, Path],
-        sr: Optional[int]=None,
-        format: Optional[str]=None,
-        mono=True
+        sr: Optional[int] = None,
+        format: Optional[str] = None,
+        mono=True,
     ) -> Union[np.ndarray, Tuple[np.ndarray, int]]:
     """
     https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI/blob/412a9950a1e371a018c381d1bfb8579c4b0de329/infer/lib/audio.py#L39
     """
-    if (isinstance(file, str) and not Path(file).exists()) or (isinstance(file, Path) and not file.exists()):
+    if (isinstance(file, str) and not Path(file).exists()) or (
+        isinstance(file, Path) and not file.exists()
+    ):
         raise FileNotFoundError(f"File not found: {file}")
     rate = 0
 
@@ -57,12 +59,25 @@ def load_audio(
     audio_stream = next(s for s in container.streams if s.type == "audio")
     channels = 1 if audio_stream.layout == "mono" else 2
     container.seek(0)
-    resampler = AudioResampler(format="fltp", layout=audio_stream.layout, rate=sr) if sr is not None else None
+    resampler = (
+        AudioResampler(format="fltp", layout=audio_stream.layout, rate=sr)
+        if sr is not None
+        else None
+    )
 
     # Estimated maximum total number of samples to pre-allocate the array
     # AV stores length in microseconds by default
-    estimated_total_samples = int(container.duration * sr // 1_000_000) if sr is not None else 48000
-    decoded_audio = np.zeros(estimated_total_samples + 1 if channels == 1 else (channels, estimated_total_samples + 1), dtype=np.float32)
+    estimated_total_samples = (
+        int(container.duration * sr // 1_000_000) if sr is not None else 48000
+    )
+    decoded_audio = np.zeros(
+        (
+            estimated_total_samples + 1
+            if channels == 1
+            else (channels, estimated_total_samples + 1)
+        ),
+        dtype=np.float32,
+    )
 
     offset = 0
 
@@ -70,8 +85,10 @@ def process_packet(packet: List[AudioFrame]):
         frames_data = []
         rate = 0
         for frame in packet:
-            frame.pts = None  # 清除时间戳，避免重新采样问题
-            resampled_frames = resampler.resample(frame) if resampler is not None else [frame]
+            # frame.pts = None  # 清除时间戳，避免重新采样问题
+            resampled_frames = (
+                resampler.resample(frame) if resampler is not None else [frame]
+            )
             for resampled_frame in resampled_frames:
                 frame_data = resampled_frame.to_ndarray()
                 rate = resampled_frame.rate
@@ -83,16 +100,21 @@ def frame_iter(container):
             yield p.decode()
 
     for r, frames_data in map(process_packet, frame_iter(container)):
-        if not rate: rate = r
+        if not rate:
+            rate = r
         for frame_data in frames_data:
             end_index = offset + len(frame_data[0])
 
             # 检查 decoded_audio 是否有足够的空间，并在必要时调整大小
             if end_index > decoded_audio.shape[1]:
-                decoded_audio = np.resize(decoded_audio, (decoded_audio.shape[0], end_index*4))
+                decoded_audio = np.resize(
+                    decoded_audio, (decoded_audio.shape[0], end_index * 4)
+                )
 
             np.copyto(decoded_audio[..., offset:end_index], frame_data)
             offset += len(frame_data[0])
+    
+    container.close()
 
     # Truncate the array to the actual size
     decoded_audio = decoded_audio[..., :offset]